[RESOLVED] Gafachi Incoming Calls

Does anyone have experience with incoming 800 numbers from Gafachi?

My current setup -

Asterisk version: 10.5.0
Freepbx version: 2.10.1.1

PEER DETAILS

type=friend
username=USERNAME
secret=PASSWORD
host=67.216.35.205
canreinvite=no
fromuser=PASSWORD
dtmfmode=rfc2833

Register => USERNAME:[email protected]

NAT settings
(from Setitngs > SIP settings page)

NAT=YES
Dynamic IP= mydyndns.com domain
localnetwork= 192.168.1.0/255.255.255.0

The numbers come in and outgoing works just fine, but incoming calls through the 800 number drop after 30 seconds. I am almost assuming that at this point it is a NAT issue or something with the firewall but I cant seem to figure it out.

Here is the CLI output of asterisk when asterisk ends the call…

[2012-07-29 15:10:46] WARNING[1658]: chan_sip.c:3686 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 103 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 31999ms with no response [2012-07-29 15:10:46] WARNING[1658]: chan_sip.c:3715 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). == Spawn extension (from-internal, *43, 4) exited non-zero on 'SIP/gafachi2a-00000002' -- Executing [[email protected]:1] Hangup("SIP/gafachi2a-00000002", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/gafachi2a-00000002'

From looking at debug mode, it seems that the my phone server is not receiving a response from gafachi, though I cannot tell which one to begin how to fix the issue.

Is there a way to stop the sending or receiving of ACK requests?

Debug output below.

<--- Reliably Transmitting (NAT) to 67.216.35.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 67.216.35.205:5060;branch=z9hG4bKdd6e9911;received=67.216.35.205;rport=5060 From: "unknown" ;tag=gss04c7296dec2ba783a01ac6ded309818c To: ;tag=as67b81fd4 Call-ID: [email protected] CSeq: 103 INVITE Server: FPBX-2.10.1(10.5.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 227

v=0
o=root 751980246 751980246 IN IP4 127.0.0.1
s=Asterisk PBX 10.5.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10106 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<— SIP read from UDP:67.216.35.205:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.35.205:5060;branch=z9hG4bK9591b26a
From: “unknown” sip:[email protected];tag=gss04c7296dec2ba783a01ac6ded309818c
To: sip:[email protected]:55084;tag=as222cd0b2
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Retransmitting #1 (NAT) to 67.216.35.205:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.35.205:5060;branch=z9hG4bKdd6e9911;received=67.216.35.205;rport=5060
From: “unknown” sip:[email protected];tag=gss04c7296dec2ba783a01ac6ded309818c
To: sip:[email protected]:55084;tag=as67b81fd4
Call-ID: [email protected]
CSeq: 103 INVITE
Server: FPBX-2.10.1(10.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 227

v=0
o=root 751980246 751980246 IN IP4 127.0.0.1
s=Asterisk PBX 10.5.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10106 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Executing [*[email protected]:2] Wait("SIP/gafachi2a-00000003", "1") in new stack

<— SIP read from UDP:67.216.35.205:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.35.205:5060;branch=z9hG4bKf6a39a4b
From: “unknown” sip:[email protected];tag=gss04c7296dec2ba783a01ac6ded309818c
To: sip:[email protected]:55084;tag=as222cd0b2
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Retransmitting #2 (NAT) to 67.216.35.205:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.35.205:5060;branch=z9hG4bKdd6e9911;received=67.216.35.205;rport=5060
From: “unknown” sip:[email protected];tag=gss04c7296dec2ba783a01ac6ded309818c
To: sip:[email protected]:55084;tag=as67b81fd4
Call-ID: [email protected]
CSeq: 103 INVITE
Server: FPBX-2.10.1(10.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 227

v=0
o=root 751980246 751980246 IN IP4 127.0.0.1
s=Asterisk PBX 10.5.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10106 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Executing [*[email protected]:3] Playback("SIP/gafachi2a-00000003", "demo-echotest") in new stack
-- <SIP/gafachi2a-00000003> Playing 'demo-echotest.ulaw' (language 'en')

<— SIP read from UDP:67.216.35.205:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.35.205:5060;branch=z9hG4bK603e7489
From: “unknown” sip:[email protected];tag=gss04c7296dec2ba783a01ac6ded309818c
To: sip:[email protected]:55084;tag=as222cd0b2
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[2012-07-29 15:13:15] NOTICE[17231]: manager.c:2461 authenticate: Seems to have passed…
Retransmitting #3 (NAT) to 67.216.35.205:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.35.205:5060;branch=z9hG4bKdd6e9911;received=67.216.35.205;rport=5060
From: “unknown” sip:[email protected];tag=gss04c7296dec2ba783a01ac6ded309818c
To: sip:[email protected]:55084;tag=as67b81fd4
Call-ID: [email protected]
CSeq: 103 INVITE
Server: FPBX-2.10.1(10.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 227

v=0
o=root 751980246 751980246 IN IP4 127.0.0.1
s=Asterisk PBX 10.5.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10106 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.35.205:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.35.205:5060;branch=z9hG4bKc3c1352c
From: “unknown” sip:[email protected];tag=gss04c7296dec2ba783a01ac6ded309818c
To: sip:[email protected]:55084;tag=as222cd0b2
Contact: sip:[email protected]
Call-ID: 20a6281e2a8db8183f143ace7[email protected]
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

I have also tried entering the user details as described by the Gafachi documentation and I still have the calls hanging up after 30 seconds…

Here are the settings they provided for the user settings.

They also state that in sip.conf we need to use [gafachi2a] so I tried to enter gafachi2a in the user context box under the trunk settings

type=friend username=USERNAME secret=PASSWORD host=67.216.35.205 canreinvite=no fromuser=USERNAME dtmfmode=rfc2833 context=from-trunk

For anyone else experiencing this issue…

Gafachi had me add insecure=invite to my peer details, now everything seems to work fine.

Now time to find out why the authentication was not working, esp since both sides received / sent / accepted the authentication.